The present invention relates to the transmission over ATM (<<Asynchronous Transfer Mode >>) networks, of data flows generated in accordance with a connected mode transport protocol and formatted into packets using a non-connected mode network protocol. In one specific application, the TCP transfer protocol (<<Transmission Control Protocol>>) and the IP network protocol (<<Internet Protocol>>) are considered.
Recent years have seen an explosion in commercial use of the Internet, both amongst the general public who have access to the Internet from home computers and from professional telecommunications services. In addition, more and more applications are appearing on the Internet networks which have stringent requirements in terms of transfer rate and quality of service (QoS).
Originally designed without any concern for quality of service and mainly geared towards data transmission, the Internet network protocols and IP in particular are based on simple and robust principles for routing packets but at the expense of operations that are costly in terms of time and software processing capacity, which can give rise to bottlenecks. As a result, the network service offered by the Internet is what is referred to as <<best effort>>, i.e. the network transfers information to the best of its ability and without any guarantee either as regards loss of information or delays in transfer. The task of restoring the integrity of transmitted data then falls to the protocols and applications embedded in the users' equipment.
Of these protocols, TCP is currently the most common because it is the transfer protocol used for exchanges linked to electronic mail (SMTP, <<Simple Mail Transfer Protocol>>), file transfers (FTP, <<File Transfer Protocol>>), Web (HTTP, <<HyperText Transfer Protocol>>), etc. In spite of the security which it affords through retransmission and flow control mechanisms, TCP does not guarantee the quality of service afforded to applications in terms of available speed and transfer time, these parameters remaining very much dependent on the status of the network at the level of the IP layer. This situation is due to the fact that the mechanisms used by TCP are only available in the terminals and do not make use of any explicit information about the resources available within the network. More specifically, the flow and congestion controls of TCP regard the network as a black box and react when the loss of packets is detected at the terminals, which leads to a significant degradation in the performance of the applications. For example, when transferring fixed images forming part of a Web page, this degradation disrupts usage (jerky displays, waiting, unrecoverable lock-up, . . . ) reflecting the level of quality of service provided by the network in a very tangible manner.
Several solutions are under study as a means of improving both speeds and quality of service accessible to Internet traffic and TCP in particular. Basically, there are two general lines of thinking as to how transmission rates can be increased:                <<all IP>> networks, which are extensions of the existing IP networks, with routers having the capacity to relay packets at high speed (giga-routers), interconnected by very high speed transmission arteries, of the SDH (<<Synchronous Digital Hierarchy>>) or WDM (<<Wavelength Division Multiplexing>>) type. Several techniques may be used to set up such routers, for example by a parallel arrangement of the packet forwarding engine within the routers or using a label switching system (the <<MultiProtocol Label Switching>> solution, or MLPS from the IETF) in order to short-circuit the process of analysing addresses packet by packet;        IP networks using the ATM technique as a switching core. The ATM network may be used directly to transfer IP traffic or, alternatively, by introducing coupling between ATM and IP to restore the concept of connection (to a more or less strict degree) in IP, for example by using the MPLS technique (<<MultiProtocol Label Switching>>) applied to the ATM.        
There are also two approaches to quality of service. The first is to improve quality of service differently from one service to another, without necessarily being able to give the user any contractual guarantee in terms of objectives. This principle is being studied by the DiffServ group of the IETF. The second approach, by contrast, is to guarantee quality of service objectives.
Several techniques have been introduced as a means of transporting TCP traffic on an ATM network:    1) Transporting Internet traffic via an ATM network having MPLS functions. ATM connections (virtual circuits or paths) are created by the MPLS functions. At the network access point, the TCP segments (encapsulated in IP packets) are directed by the MPLS edge router to one or other of the established ATM connections depending on the destination IP address. Several TCP connections may be grouped on a same virtual circuit (<<VC>>). Furthermore, several virtual circuits at the input of a network element can be merged to a same output circuit (<(VC merging >>). The major problem of MPLS in its current version is that it lacks the tools to manage traffic. The flows manipulated by MPLS are of the UBR type (<<Unspecified Bit Rate>>) and are therefore lacking in quality of service. Attempts are currently being made to combine the MPLS and DiffServ approaches. This can be easily resolved by introducing priority mechanisms between flows at the output ports of the switches/routers. Commercially available products such as <<(PacketStar>> sold by the company Lucent are already offering an approach of this type.    2) Transporting TCP traffic via generic ATM transfer capacities, namely:            SBR: <<Statistical Bit Rate>>), also referred to as <<Variable Bit Rate >> (VBR) at the ATM Forum;        ABR: <<Available Bit Rate>>;        UBR: <<Unspecified Bit Rate>>, optionally with mechanisms to discard packets selectively (EPD, <<Early Packet Discard>>, or PPD, <<Partial Packet Discard>>);        GFR: Guaranteed Frame Rate >>.Statistical traffic parameters (e.g. sustainable speed and maximum burst size in the case of SBR capability) are determined a priori. In order to transport TCP traffic, a statistical ATM connection is established which must comply with the prescribed traffic contract. Solutions falling within this family are inadequate because generic ATM transfer capabilities require fixed traffic parameters (speed and associated cell volume) whereas the TCP control mechanism is based only on the volumes of information transferred and does not take account of speed as a parameter. Furthermore, the volume of information transmitted by TCP depends on traffic conditions within the network and hence a large number of parameters which are uncontrollable as a whole and which fluctuate over time. It is therefore difficult to determine statistical traffic parameters to qualify the TCP flows. If the parameters chosen are inaccurate, this can lead to losses in the font mechanisms at the network input, thereby causing a significant deterioration in the transfer quality of the TCP flows.            3) TCP regulation by speed control (ACK bucket, control of window size). Observations (estimation of transfer time) are made in order to deduce the state of congestion on the network and adapt the dynamics of TCP by regulating the transmission of the acknowledgement segments to avoid losing information in the network and to set a speed value for TCP. This technique is applicable even if the transport network is not of the ATM type. This TCP adjustment by controlling speed is based on adaptive and empirical flow control methods. They are applied at the periphery of the network which nevertheless remains basically of the <<best effort >> type. In particular, resources are not reserved in the different nodes of the network and it is therefore not possible for the network to provide any contractual guarantee with regard to quality of service objectives.    4) Coupling TCP with broadband signalling. Whenever a TCP connection has to be made, an Internet signalling message (for example RSVP) is issued by the source terminal and a conventional broadband signalling procedure (complying with the Q.2931, Q.2963 protocols, etc. of the UIT-T) is initiated by the router at the network input in order to establish an ATM connection corresponding to the TCP connection. Unfortunately, the response times in broadband signalling are too long for TCP transport because of the processing which has to be applied by the software to the signalling messages (which vary in size and contain too much non-relevant information in the case of TCP streams) in the signalling controllers and of various security protocols used when exchanging signalling messages (SSCOP). It has now been more or less admitted that broadband signalling is not suitable for transactions taking place on the Internet.    5) Proposals for lightweight signalling. On the grounds that the procedure of coupling TCP and broadband signalling described above is too laborious and too slow, proposals have recently appeared in various publications as to how the signalling procedures can be simplified, and in particular what are known as the UNITE systems (see G. Hjámtýsson et al., <<UNITE—An Architecture for Lightweight Signaling in ATM Networks>>, Proc. Infocom'98, New York, April 1998), OPENET, (see I. Cidon et al. <<OPENET: An Open and Efficient Control Platform for ATM Networks >>, Proc. Infocom'98, New York, April 1998) and Dynaflow (see Q. Bian et al., <<Dynamic Flow Switching—A New Communication Service for ATM Networks>>, Proc. Infocom'98, New York, April 1998).            In UNITE, proposed by the company AT&T, detection of a stream prompts a mono-cell signalling procedure initialised in the network in order to set up an ATM VC. The establishment is made hop by hop and includes setting up the connection and allocating resources.        In OPENET, proposed by the company Sun Microsystems, conventional broadband signalling is used between the user and the network. Within the network, the routes are established by P—NNI and the route activation is performed by means of a mono-cell signalling scheme.        In DYNAFLOW, proposed by the George Washington University, the IP packets segmented into ATM cells are regrouped in the form of datagrams with a specific header for resources management, the purpose of which is to switch the datagram into the network and reserve the resources for the datagram.            6) Open loop multiplexing and congestion control using techniques to selectively discard TCP Packets (PPD, EPD, WRED, etc.) and mechanisms to assign Priority between traffic classes. Given that TCP reacts to loss by reducing the volume of information transmitted, the simplest way of transporting TCP flows is to allow them into the network without any control of statistical parameters and to multiplex them in open loop. Some packets will then be entirely (EPD) or partially (PPD) discarded if the buffer memories in the network become full. This approach can be refined by introducing priority mechanisms between flows: certain flows can be delivered to the output ports of the network elements more rapidly than others (<<Expedited Forwarding >>); or a larger memory space might be reserved for certain flows to guarantee that less information is lost (<<Assured Forwarding>>). These open loop statistical multiplexing schemes which use priority mechanisms and techniques to discard TCP packets selectively are not capable of guaranteeing quality of service objectives. They may be used to selectively improve the transfer quality of one class of service as opposed to another but are in no way capable of guaranteeing objectives.
It has been shown that a mechanism for spacing ATM cells could be used to regulate the dynamics of TCP on an ATM network (see F. Guillemin et al., <<Regulation of TCP over ATM via Cell Spacing>>, Proc. ITC'16, Edinburgh, June 1999). A spacing mechanism of this type may also prevent buffer memories of the network from overflowing in the case of the non-connected mode UDP protocol (<<User Datagram Protocol>>). On this basis, a lightweight signalling protocol has been developed, known as ASIA, which associates each transfer of Internet information, via TP or UDP, with an ATM virtual circuit of the ABT type (<<ATM Block Transfer>>), including the establishment and definition of resources within the network by exchanging mono-cell messages made up of resource management (RM) cells (see J. Boyer et al., <<Accelerated Signaling for the Internet over ATM (ASIA)>>, European Transactions on Telecommunications, 1999, Special issue on architectures, protocols and quality of service for the Internet of the future).
An object of the invention is to guarantee objectives (in terms of information loss, minimal usable bandwidth and/or transfer time) in transfer quality of the traffic of a connection of a transport protocol such as TCP/IP carried by an ATM connection.